Druid: Open Source Unified Communications Based on Asterisk

A group called Voiceroute just pushed out a product called Druid (actually two products, an open source version and a commercial version) that looks very intriguing. It is based on Asterisk at the core but they've done a lot of work on top of it -- besides just sticking on a useful GUI. This is what I love about the telephony space today -- continuous improvement as folks figure out what they really want. There's a decent little preview/review here. A more thorough explanation of features here. Some ideas for applications here.

So many forms of communications (voice, voicemail, email, IM, mobile, fax), so many ways to leverage it. Now a single IP communications platform based on open source and open standards.

What is Druid? Druid is the premier unified communications platform for enterprises. It allows companies to deploy easily and affordable high endIP communications services using off the shelve commodity hardware and IP phones. Druid covers your enterprise communication needs from IP voice, voicemail, IM all the way to the mobile space.

I love Asterisk and the various things which can be be built with it. But, at its core, (stock) Asterisk excels most at being an engine at the center of a telephony/communications platform. To maximize adoption, the barrier to use must be lowered through improved management interfaces and turnkey features out of the box that reflect what end-users want. There's lots of room for improvement for specific problem domains. That's why Asterisk's flexibility is so powerful and having a lively eco-system around it makes it more and more accessible to a broader base of users. That doesn't mean that Asterisk based solutions aren't ready for prime time today -- they most definitely are. Thankfully, Asterisk's eco-system is strong with solutions such as Fonality, FreePBX, Digium's AsteriskNOW/ABE/AA50, and many others. Oh, and now we have Voiceroute's Druid.

Definitely will be taking a look Druid soon.

Improvements at Digium? ....and New AA50 Firmware

Yesterday I noted that Digium pushed out a new firmware release for their small business targeted hardware appliance (the AA50) and it looks like a good sized update. It is dated the 4th in the release notes but the 14th elsewhere so I think one or the other is a typo.

They appear to be syncing it up with their other code bases (Asterisk Business Edition, AsteriskNOW). From what I've seen in the public commits and hints on mailing lists from staff, they've been hitting pretty hard on both of those in the QA department of late.

Digium is clearly working on changing how they manage their code bases and the mixture of their commercial and open source releases, to address different customer/user segments and some of the problems that have come up. There was even a note about a split release change they are going to with AsteriskNOW for the GPL purists. Glad to see some serious re-thinking about releases, meeting the desires of different customers, quality control enhancements, and the public evidence of the aforementioned outside of press releases.

I imagine things will continue to trickle down and we'll see more improvements coming together. I'm sure some will turn out to be good ideas while others head back to the drawing board but that's how most progress works, eh? :)

The additional leverage should improve things across the board no matter where you fall in the Asterisk eco-system. Even if you don't use an Asterisk based telephony solution, their moves impact the others in the marketplace (and visa-versa).

Back to the new AA50 firmware release: It would appear to address IVR and ring group related bugs I encountering early on in the GUI, that I ended up bypassing the GUI to workaround. I haven't tried it yet on the AA50 I have access to. I'll probably give it another couple o' weeks in the wild first since it's a .0.1 rev. That client has some other problems and improvements they want me to attack first anyhow. Below are change log excerpts:

Version 1.1.0.1 - February 4, 2008
* Enable Internationalization settings in the GUI
* Provide the ability to select between kewlstart and loopstart
* g722 codec is available
* WAN Side Provisioning of Polycom phones is enabled
* AA50 has been synchronized to ABE C.1 branch source code
* Ring groups number of seconds field has been added
* Ring group bug using ivr option has been fixed.
* Polycom bug concering use of standard timezone has been fixed
* Blackfin math for meetme conferencing has been implemented
* Adjustable flash hook duration is now available in teh GUI
* Calling rule editing and deleting bug has been fixed.
* Bug concerning setting incoming rule to choose voicemail is fixed.
* DTMF twist settings for Brazil have been added to tone generation
* Call forwarding loops have been prevented so that they do not crash the AA50
* Voicemail attachments are set to WAV format.
* Network setting tabs no longer disappear.
* Added an optional full-wave mode DAA ring detect in the sx00i driver